maybe you need to clean the iaxmodem sources including the lib to be able to recompilemake cleanthen try agin ./build staticOn 8/17/06, Giedrius Augys
<[EMAIL PROTECTED]> wrote:Hi , I removed from /usr/local all spandsp and iax libraries and ./build static . But I get this:
creating libspandsp.la(
what's the use of enabling multiple codecs for a peer?
can asterisk avoid trancoding when both phones are capable of a common codec enabled for them?
e.g. first priority codec = g729, 2nd= ulaw,,, if phone 1 calls another
ip phone capable of g729 to use it and when calling through Zap, can
asterisk
i used to have this problem,
with me, it appeared that i had to press the feature keys very quickly.
my solution was to set featuredigittimeout higher than the default 500.
also, when i use IAX phones, i had to set dtmf to ouband audio for asterisk to recognize the keys pressed.On 4/28/06, Ronald W
i am also getting this warning since upgrading to 1.2 when running asterisk with -p param (realtime priority)On 4/6/06, Raymond Chen <
[EMAIL PROTECTED]> wrote:Tomislav Parčina wrote:> In article <
[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...>>> Hi What causes deadlock? Apr 5 14:02:43 WAR
hi group,
is there a way that SIP phones be allowed to use G.729 passthrough when
calling each other and when calling PSTN through Zap that asterisk
force the phones to use ulaw.
thanks,
ultor
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try this:
http://www.oinko.net/astrecipes/index.php?n=152
On 3/22/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
On 3/22/2006, "John Novack" <[EMAIL PROTECTED]
> wrote:>> Douglas Garstang wrote:>>> Good grief. Considering all the libraries and requirements, it'd almost>> be easier to find some
hi,
any experienced user please guide us newbies to tune echo training intelligently.
echocancel can have values of 32,64,128,256, and echotraining=400 or 800
my test setup: SoftFone --- * --- PSTN
from the default values of echocancel=128 and training=400, i observed
that if SoftFo
